Method and apparatus for automatically recognizing input audio and/or video streams

ABSTRACT

A method and system for the automatic identification of audio, video, multimedia, and/or data recordings based on immutable characteristics of these works. The invention does not require the insertion of identifying codes or signals into the recording. This allows the system to be used to identify existing recordings that have not been through a coding process at the time that they were generated. Instead, each work to be recognized is “played” into the system where it is subjected to an automatic signal analysis process that locates salient features and computes a statistical representation of these properties. These features are then stored as patterns for later recognition of live input signal streams. A different set of features is derived for each audio or video work to be identified and stored. During real-time monitoring of a signal stream, a similar automatic signal analysis process is carried out, and many features are computed for comparison with the patterns stored in a large feature database. For each particular pattern stored in the database, only the relevant characteristics are compared with the real-time feature set. Preferably, during analysis and generation of reference patterns, data are extracted from all time intervals of a recording. This allows a work to be recognized from a single sample taken from any part of the recording.

This is a continuation of U.S. patent application No. 11/672,294, filedFeb. 7, 2007, now U.S. Pat. No. 7,870,574, which is a continuation ofU.S. patent application No. 09/420,945, filed Oct. 19, 1999 now U.S.Pat. No. 7,194,752, which claims priority benefit of U.S. patentapplication No. 60/155,064, filed Sept. 21, 1999, both of which areincorporated herein by reference.

BACKGROUND OF THE INVENTION

1. Technical Field

The present invention relates to apparatus and method for automaticallyrecognizing signals, particularly audio and video signals that may betransmitted via broadcast, computer networks, or satellite transmission.This has particular application in the detection of the transmission ofcopyright-protected material for royalty payment justification, and inthe verification of transmission of scheduled programming andadvertising.

2. Related Art

The need for automatic recognition of broadcast material has beenestablished, as evidenced by the development and deployment of a numberof automatic recognition systems. The recognized information is usefulfor a variety of purposes. Musical recordings that are broadcast can beidentified to determine their popularity, thus supporting promotionalefforts, sales, and distribution of media. The automatic detection ofadvertising is needed as an audit method to verify that advertisementswere, in fact, transmitted at the times and for the duration that theadvertiser and broadcaster agreed upon. Identification ofcopyright-protected works is also needed to assure that proper royaltypayments are made. With new distribution methods, such as the Internetand direct satellite transmission, the scope and scale of signalrecognition applications has increased.

Automatic program identification techniques fall into the two generalcategories of active and passive. The active technologies involve theinsertion of coded identification signals into the program material orother modification of the audio or video. Active techniques are facedwith two difficult problems. The inserted codes must not causenoticeable distortion or be perceivable to listeners and viewers.Simultaneously, the identification codes must be sufficiently robust tosurvive transmission system signal processing. Active systems that havebeen developed to date have experienced difficulty in one or both ofthese areas. An additional problem is that almost all existing programmaterial has not yet been coded. The identification of these works istherefore not possible. For this reason we will dismiss the activetechnologies as inappropriate for many important applications.

Passive signal recognition systems identify program material byrecognizing specific characteristics or features of the signal. Usually,each of the works to be identified is subjected to a registrationprocess where the system “learns” the characteristics of the audio orvideo signal. The system then uses pattern-matching techniques to detectthe occurrence of these features during signal transmission. One of theearliest examples of this approach is presented by Moon et al. in U.S.Pat. No. 3,919,479 (incorporated herein by reference). Moon extracts atime segment from an audio waveform, digitizes it and saves thedigitized waveform as a reference pattern for later correlation with anunknown audio signal. Moon also presents a variant of this techniquewhere low bandwidth amplitude envelopes of the audio are used instead ofthe audio itself. However, both of Moon's approaches suffer from loss ofcorrelation in the presence of speed differences between the referencepattern and the transmitted signal. The speed error issue was addressedby Kenyon et al. in U.S. Pat. No. 4,450,531 (incorporated herein byreference) by using multiple segment correlation functions. In thisapproach, the individual segments have a relatively low time-bandwidthproduct and are affected little by speed variations. Patterndiscrimination performance is obtained by requiring a plurality ofsequential patterns to be detected with approximately the correct timedelay. This method is accurate but somewhat limited in capacity due tocomputational complexity.

A video program identification system is described by Kiewit et al. inU.S. Pat. No. 4,697,209 (incorporated herein by reference). This systemdetects events such as scene changes to identify program changes. When achange is detected, a signature is extracted from the video signal andstored along with the time of occurrence. A similar process is performedat a central location for each available program source. Periodicallythe central site interrogates the stored data at the viewer location toobtain the signatures. These are compared to identify the changedprogram selection. This method has the advantage of only needing toselect among a limited set of possibilities, but has the disadvantagethat the queuing events that trigger signature extraction are notparticularly reliable.

Another video recognition system is described by Thomas et al. in U.S.Pat. No. 4,739,398 (incorporated herein by reference). The methoddiscussed by Thomas identifies video programs by matching video featuresselected from a number of randomly selected locations in the framesequence. The intensity, etc. of each location is quantized to one bitof resolution, and these bits are stored in a single word. A sequence offrame signatures is acquired from a program interval with the spacing offrame signatures selected according to a set of rules. Noisy or errorprone bits within the signature words are masked. In the preferredembodiment there are eight frame signatures per interval each containingsixteen binary values. A key word is chosen from the frame signature setand is used to stage the pattern recognition process. When the key wordis detected by bit comparison, a table of candidate patterns is accessedto locate a subset of patterns to be evaluated. These templates are thencompared with the current video signature. Audio recognition ismentioned but no method is presented. Thomas also describes methods forcompressing audio and video signals for transmission to a centrallocation for manual identification. Corresponding video signatures arealso transmitted. This allows the acquisition of unknown programmaterial so that the new material can be added to a central library forlater identification. The unknown signatures transmitted from the remotesites can be identified from templates stored in the central library orby manual viewing and listening to the corresponding compressed videoand audio.

An audio signal recognition system is described by Kenyon et. al in U.S.Pat. No. 4,843,562 (incorporated herein by reference) that specificallyaddresses speed errors in the transmitted signal by re-sampling theinput signal to create several time-distorted versions of the signalsegments. This allows a high-resolution fast correlation function to beapplied to each of the time warped signal segments without degrading thecorrelation values. A low-resolution spectrogram matching process isalso used as a queuing mechanism to select candidate reference patternsfor high-resolution pattern recognition. This method achieves highaccuracy with a large number of candidate patterns.

In U.S. Pat. No. 5,019,899 Boles et al. (incorporated herein byreference) describe a video signal recognition system that appears to bea refinement of the Thomas patent. However, the method of featureextraction from the video signal is different. After digitizing a frame(or field) of video, the pixels in each of 64 regions is integrated toform super-pixels representing the average of 16×16 pixel arrays.Thirty-two pairs of super-pixels are then differenced according to apredefined pattern, and the results are quantized to one bit ofresolution. As in the Thomas patent, a program interval is representedby eight frame signatures that are selected according to a set of rules.The pattern matching procedure involves counting the number of bits thatcorrectly match the input feature values with a particular template.Boles also presents an efficient procedure for comparing the unknowninput with many stored templates in real-time. For purposes of thisinvention, real-time operation requires all patterns to be evaluated ina thirtieth of a second.

Lamb et al. describe an audio signal recognition system in U.S. Pat. No.5,437,050 (incorporated herein by reference). Audio spectra are computedat a 50 Hz rate and are quantized to one bit of resolution by comparingeach frequency to a threshold derived from the corresponding spectrum.Forty-eight spectral components are retained representing semitones offour octaves of the musical scale. The semitones are determined to beactive or inactive according to their previous activity status andcomparison with two thresholds. The first threshold is used to determineif an inactive semitone should be set to an active state. The secondthreshold is set to a lower value and is used to select active semitonesthat should be set to an inactive state. The purpose of this hysteresisis to prevent newly occurring semitones from dominating the powerspectrum and forcing other tones to an inactive state. The set of 48semitone states forms an activity vector for the current sampleinterval. Sequential vectors are grouped to form an activity matrix thatrepresents the time-frequency structure of the audio. These activitymatrices are compared with similarly constructed reference patternsusing a procedure that sums bit matches over sub-intervals of theactivity matrix. Sub-intervals are evaluated with a several differenttime alignments to compensate for speed errors that may be introduced bybroadcasters. To narrow the search space in comparing the input withmany templates, gross features of the input activity matrix arecomputed. The distances from the macro features of the input and eachtemplate are computed to determine a subset of patterns to be furtherevaluated.

In U.S. Pat. No. 5,436,653 Ellis et al. (incorporated herein byreference) discuss a technique that seems to be a derivative of theThomas and Boles patents. While the super-pixel geometry is differentfrom the other patents, the, procedures are almost identical. As in theBoles patent, super-pixels (now in the shape of horizontal strips) indifferent regions of a frame are differenced and then quantized to onebit of resolution. However, sixteen values are packed into a sixteen-bitword as in the Thomas patent, representing a frame signature.Potentially noisy bits in the frame signature may be excluded from thecomparison process by use of a mask word. Frames within a programinterval are selected according to a set of rules. Eight framesignatures of sixteen bits each are used to represent a programinterval. As in the Thomas patent, one of the frame signatures isdesignated as a “key signature”. Key signature matching is used as aqueuing mechanism to reduce the number of pattern matching operationsthat must be performed in the recognition process. Ellis addressesclumping of patterns having the same key signature as well as videojitter that can cause misalignment of superpixels. In addition, Ellisdescribes a method of using multiple segments or subintervals similar tothe method described in the Kenyon et al. 4,450,531 patent. Unlike theThomas and Boles patents, Ellis offers an audio pattern recognitionsystem based on spectrogram matching. Differential audio spectra arecomputed and quantized to form sixteen one-bit components. Groups ofthese spectral signatures are selected from a signal interval. Ellis hasupdated this method as described in U.S. Pat. No. 5,621,454(incorporated herein by reference).

Forbes et al. describe in U.S. Pat. No. 5,708,477 (incorporated hereinby reference) a system that is used to automatically edit advertisementsfrom a television signal by muting the television audio and pausing anyVCR recording in progress. This is done by first detecting changes inthe overall brightness of a frame or portion of a frame indicating ascene change. When a scene change is detected, a lowpass filteredversion of the frame is compared with a similar set of frames that havebeen previously designated by the viewer to indicate the presence of anadvertisement. When a match is detected, the audio/video is interruptedfor an amount of time specified by the viewer when the segment wasdesignated by the viewer as an advertisement. The detection decision isbased on a distance metric that is the sum of the absolute values ofcorresponding input and template region differences. The intensity ofvarious regions appears to be computed by'averaging video scan lines.Forbes does not use any audio information or time series properties ofthe video.

While the inventions cited above in the prior art indicate progress inthe technical field of automatic signal identification, there are anumber of shortcomings in these technologies. To be accepted in themarketplace a system must have sufficient processing capacity to searchsimultaneously for a very large number of potential patterns from manydifferent sources. The technologies of the prior art underestimate themagnitude of this capacity requirement. Further, if the capacity of theprior art systems is increased in a linear fashion through the use offaster processors, recognition accuracy problems become evident. Theseproblems are in part due to the underlying statistical properties of thevarious methods, but are also caused by intolerance of these methods tosignal distortion that is typical in the various media distribution andbroadcast chains. Most of the cited inventions are designed to handleeither audio or video but not both. None of the inventions in the priorart are capable of blending audio and video recognition in a simpleuniform manner. While the duration of samples required for recognitionvaries among the different techniques, none of them is capable ofrecognizing a short segment from any part of a work and the moving to adifferent channel.

Thus, what is needed is a signal recognition system that can passivelyrecognize audio and/or video data streams in as little as six secondswith great accuracy. Preferably, the system can recognize any portion ofthe input data stream, thus allowing channel-hopping as the systemquickly recognizes one broadcast work and moves on to another.

SUMMARY OF THE INVENTION

It is an object of the present invention to overcome the problems andlimitations described above and to provide a system for recognizingaudio, video, mixed, and/or data streams with great accuracy, minimaltime, and with fewer processors.

In one aspect of the present invention, an apparatus for recognizing aninput data stream comprises a receiver for receiving the input datastream, and an interface for (i) randomly selecting any one portion ofthe received data stream and (ii) forming a first plurality of featuretime series waveforms corresponding to spectrally distinct portions ofthe received data stream. A memory is provided for storing a secondplurality of feature time series waveforms. One or more processors areprovided for correlating the first plurality of feature time serieswaveforms with the second plurality of feature time series waveforms,and for designating a recognition when a statistic of correlation valuesbetween the first plurality of feature time series waveforms and one ofthe second plurality of feature time series waveforms reaches apredetermined value.

According to another aspect of the present invention, an apparatus forforming video features from an input stream includes a receiver forreceiving an input video stream which corresponds to a video screenhaving a plurality of regions, the video screen comprising a pluralityof pixels having intensity and color. One or more processors areprovided for (i) receiving the video stream from the receiver, (ii)integrating at least one of the intensity and the color of video signalscorresponding to each of the plural areas of the video screen, (iii)forming a set of low rate time series data streams from the integratedvideo signals, (iv) forming overlapping time intervals of the multiplefeature streams such that the overlapping time intervals encompass theentire received video frame sequence, (v) determining the mostdistinctive information from each time interval, (vi) rank-ordering thetime interval segments according to their distinctness, (vii)transforming the rank-ordered time interval segments to produce complexspectra, and (viii) storing the formed data as video features.

According to yet another aspect of the present invention, apparatus forforming audio features from an input audio stream has a receiver forreceiving the input audio stream and separating the received audiostream into a plurality of different frequency bands. Processorstructure is included for (i) extracting energy from each of theplurality of frequency bands, (ii) integrating the energy extracted fromeach of the plurality of frequency bands, (iii) forming multiple featurestreams from the integrated energy, (iv) forming overlapping timeintervals of the multiple feature streams such that the overlapping timeintervals encompass the entire received audio stream, (v) determiningthe most distinctive information from each time interval, (vi)rank-ordering the time interval segments according to theirdistinctness, and (vii) transforming the rank-ordered time intervalsegments to produce complex spectra. A memory is also provided forstoring the transformed complex spectra.

In another aspect of the present invention, a method for recognizing aninput data stream, comprises the steps of: (i) receiving the input datastream; (ii) randomly selecting any one portion of the received datastream; (iii) forming a first plurality of feature time series waveformscorresponding to spectrally distinct portions of the received datastream; (iv) storing a second plurality of feature time serieswaveforms; (v) correlating the first plurality of feature time serieswaveforms with the second plurality of feature time series waveforms;and (vi) designating a recognition when a correlation probability valuebetween the first plurality of feature time series waveforms and one ofthe second plurality of feature time series waveforms reaches apredetermined value.

In still another aspect of the present invention, a method for formingvideo features from an input video stream, comprises the steps of: (i)receiving an input video stream which corresponds to a video screenhaving a plurality of regions, the video screen comprising a pluralityof pixels having intensity and color; (ii) integrating at least one ofthe intensity and the color of video signals corresponding to each ofthe plural areas of the video screen; (iii) forming a set of low ratetime series data streams from the integrated video signal; (iv) formingoverlapping time intervals of the multiple feature streams such that theoverlapping time intervals encompass the entire received audio stream;(v) determining the most distinctive information from each timeinterval; (vi) rank-ordering the time interval segments according totheir distinctness; (vii) transforming the rank-ordered time intervalsegments to produce complex spectra; and (viii) storing the transformedcomplex spectra as video features.

According to a further aspect of the present invention, a method forforming audio features from an audio stream, comprises the steps of: (i)receiving the input audio stream and separating the received audiostream into a plurality of different frequency bands; (ii) extractingenergy from the plurality of frequency bands; (iii) integrating theenergy extracted from each of the plurality of frequency bands; (iv)forming multiple feature streams from the integrated energy; (v) formingoverlapping time intervals of the multiple feature streams such that theoverlapping time intervals encompass the entire received audio stream;(vi) determining the most distinctive information from each timeinterval; (vii) rank-ordering the time interval segments according totheir distinctness; (viii) transforming the rank-ordered time intervalsegments to produce complex spectra; and (ix) storing the transformedcomplex spectra as audio features.

In a further aspect of the present invention, a computer readablestorage medium stores a program which causes one or more computers torecognize an input data stream, the stored program causing the one ormore computers to: (i) receive the input data stream; (ii) randomlyselect any one portion of the received data stream; (iii) form a firstplurality of feature time series waveforms which corresponds tospectrally distinct portions of the received data stream; (iv) store asecond plurality of feature time series waveforms; (v) correlate thefirst plurality of feature time series waveforms with the secondplurality of feature time series waveforms; and (vi) designate arecognition when a correlation probability value between the firstplurality of feature time series waveforms and one of the secondplurality of feature time series waveforms reaches predetermined value.

According to yet another aspect of the present invention, a method forforming recognition features from an input data stream, comprises thesteps of: (i) receiving the input data stream; (ii) forming a pluralityof feature time series waveforms which correspond to spectrally distinctportions of the received input data stream; (iii) forming multiplefeature streams from the plurality of feature time series waveforms;(iv) forming overlapping time intervals of the multiple feature streams;(v) estimating the distinctiveness of each feature in each timeinterval; (vi) rank-ordering the features according to theirdistinctiveness; (vii) transforming the feature time series to obtaincomplex spectra; and (viii) storing the feature complex spectra as therecognition features.

BRIEF DESCRIPTION OF THE DRAWINGS

The above objects and other advantageous features of the presentinvention will be more readily understood from the following detaileddescription of the preferred embodiment when taken in conjunction withthe drawings which are described below.

FIG. 1 illustrates a system level diagram of the signal recognitionsystem. The system includes one or more Audio/Video Interface Subsystemsthat collect signals and extract feature packets to be identified. Alsoincluded are one or more Pattern Recognition Subsystems that perform theactual signal identifications. One or more Pattern InitializationSubsystems are included that generate the reference patterns fromrecorded media. These patterns are stored in a Master Pattern Databaseas well as being distributed to the Pattern Recognition Subsystems. ASearch Robot and Scheduler locates signals of interest and controlsacquisition of feature packets and distribution of these packets amongPattern Recognition Subsystems. Detections and their time of occurrenceare stored in a Management Database System for the production ofreports. The system is organized as a network and is also connected tothe Internet to allow access to online distribution sites.

FIG. 2 is a block diagram of the Audio Interface and Signal Processor.This is a component of the Audio/Video Interface Subsystem. There aretypically several of these boards in each Interface Subsystem. Eachboard connects to many audio sources and produces feature packets to beidentified.

FIG. 3 shows the signal processing functions that are performed toextract multiple low-bandwidth feature streams from each of severalaudio streams. These processes include repetitive spectral analyses andthe estimation of energy in each of several frequency bands. Sequencesof these energy estimates are then lowpass filtered and decimated toproduce low speed feature time series.

FIG. 4 illustrates a typical audio power spectrum and the partitioningof this spectrum into several frequency bands. Lower frequency bands arenarrower than the higher frequency bands to balance the total power ineach band and to match the human auditory characteristics.

FIG. 5 is a block diagram of the Video Interface and Signal Processor.This is also a component of the Audio/Video Interface Subsystem andproduces low bandwidth feature packets from video data. There aretypically several of these boards in each Interface Subsystem. Eachboard connects to several video sources and processes multiple videostreams in real-time.

FIG. 6 is a description of the signal processing functions used toproduce video feature time series data. For each video frame we extracta measurement frame that may consist of spatial characteristics such asintensity, color, or texture. A spatial weighting function is thenapplied to multiple regions of the frame and the energy in each regionis integrated. The integrated energy in each region is then sampled toform multiple feature time series streams. Each stream is then lowpassfiltered and decimated to produce low sample rate video features.

FIG. 7 is an example of the regions from which the video features areextracted. These regions have been selected so that reliable featurescan be extracted from video frames with very coarse spatial resolutionand any of a number of aspect ratios.

FIG. 8 is an example of the video spatial weighting function that isapplied to each region. The effect of this window is to weight pixelsnear the center of the region more heavily than those near the edges.This reduces the sensitivity of the features to spatial translation andscaling errors.

FIG. 9 illustrates several feature time series waveforms.

FIG. 10 illustrates the partitioning of a single feature waveform intooverlapped segments. These segments are then normalized, processed, andstored in the pattern database for later recognition.

FIG. 11 shows the signal processing steps that are used to generate areference pattern data structure from the feature time series waveforms.First the features from the entire work are grouped into a sequence ofoverlapping time segments. Each feature from each segment is then blockscaled to a fixed total power. The scaled feature is then processed by afast Fourier transform to produce the complex spectrum. The slidingstandard deviation of the scaled feature is also computed over aninterval equal to half of the segment length. The individual datastructures representing each feature of each segment are thenconstructed. When all features of all segments have been processed, thefeatures within each segment are rank ordered according to theirinformation content. The top level of the pattern data structure is thenconstructed.

FIG. 12 illustrates the structure of a database reference pattern entry.A reference pattern identification code is used to both the referencepattern data structures and a data structure that describes the work.The reference pattern data structure identifies whether the work isaudio or video or both. It then includes a list of pointers to segmentdescriptors. Each segment descriptor contains pattern and segmentidentification codes and a list of pointers to feature structures. Eachfeature structure contains pattern, segment, and feature identificationcodes and the pattern data itself.

Included in the pattern data are the scale factor used to normalize thedata, the standard deviation of random correlations, a detectionprobability threshold, and a rejection probability threshold. Afterthese parameters are the complex spectrum of feature waveform and thesliding standard deviation (RMS) of the feature waveform. Each componentof the overall data structure may also contain a checksum to validatedata integrity.

FIG. 13 is an example of the channel and pattern scheduler. If the audioand/or video on the current channel is known from a previous recognitioncycle, the channel can be removed from the schedule for the amount oftime remaining in a particular work. This is determined from the segmentnumber identified and the total number of segments in the work. Thesignal input can then be switched to a different source to sample andidentify its content. Depending on the duration of a particular work thesystem must search for it at different intervals. For example, briefadvertisements must be checked on each update cycle while the systemcould check for feature length movies at intervals of several minutes.This is accomplished by grouping patterns into several lists accordingto their duration. In the figure three lists are shown. The systemprocesses only a part of the longer lists during each update cycle toconserve computational resources. Once detection results have beenreported another input channel is selected and the process is repeated.Note that all of these processes will normally be occurring in parallelon several different processors for many channels and many patterns.

FIG. 14 is the preprocessing of features that occurs during real-timepattern recognition. A new block of feature data is acquired and themean is removed from each feature. Each feature is then normalized tofixed total power. The normalized feature blocks are then padded todouble their length by appending zeros. The fast Fourier transform ofeach feature block is then computed to produce the complex spectrum.

FIG. 15 shows the strategy and procedure used to identify a work using asubset of available features. The unknown input feature block iscompared with each segment of each work. For each segment of a workfeatures are evaluated sequentially according to their informationcontent. The probability of false alarm is estimated each time newinformation is added. Detection/rejection decisions are made on thebasis of two sets of probability thresholds.

FIG. 16 illustrates the feature correlation process between an unknownfeature complex spectrum and a candidate reference pattern complexspectrum. The cross-power spectrum is first computed prior to computingthe inverse FFT, yielding a cross-correlation function. The first halfof this is normalized by the sliding standard deviation. The second halfof the correlation functions contains circularly wrapped values and isdiscarded.

FIG. 17 is an example of a feature correlation function containing adetection event.

FIG. 18 illustrates how false detection probabilities are derived from adistribution of random correlation values. As shown in (A), theprobability density function of mismatched correlation values isestimated for a large group of background patterns duringinitialization. The cumulative distribution function (B) is thenestimated by integrating (A). Finally, the probability of false alarm isestimated by subtracting the CDF from one as shown in (C).

DETAILED DESCRIPTION OF THE PRESENTLY PREFERRED EXEMPLARY EMBODIMENT

1. Introduction

The preferred embodiment of the present invention is a highly flexiblesignal collection and identification system that is capable ofprocessing audio, video, multimedia signals, data signals, etc. fromdiverse sources. These sources include conventional broadcast, satellitedistribution feeds, Internet, data distribution networks, and varioussubscription services. To accomplish these objectives, the preferredexemplary embodiment is configured as a distributed network of computersubsystems where each subsystem has specific functions. These subsystemscan be replicated as necessary to provide the needed number of inputsand support the recognition of as many different works as desired. Forexample, one broadcast audio and video signal recognition station in onecity may comprise one multi-channel video receiver, one multi-channelaudio receiver, six audio interface computers, six video interfacecomputers, one scheduler computer, and a mass data storage device. Eachof the computers may comprise a Pentium CPU with appropriate RAM anddisk storage, digital signal processors, and standard LAN and Internetconnections. Of course, each recognition station may be configured withthe appropriate hardware and software to detect those signals, which areof interest at that station.

2. System Overview

The present invention discloses a technology and system for theautomatic identification of signals using a method known as passivepattern recognition. The method described is capable of identificationof program material based on audio content, video image sequences, or acombination of both. As contrasted with active signal recognitiontechnology, which injects identification codes into the recordedmaterial, the passive approach uses characteristics or features of therecording itself to distinguish it from other possible audio or videoinputs. While both methods have their advantages, passive approaches aremost appropriate for copyright management and monitoring. There areseveral reasons for this. First, coded identification signals that areadded to the audio or video material in active systems are frequentlydetectable to the discerning eye or ear. When the code injection levelis reduced to the point that it is invisible or inaudible, thereliability of the code recovery suffers. Further, the injected codesare often destroyed by broadcast processing or signal processingnecessary to distribute audio and video on computer networks. However,the most important shortcoming of the active technologies is that thereare millions of works in distribution that have not been watermarked.This material cannot be protected; only new releases that have beenprocessed to inject codes can be detected automatically using activetechniques.

In contrast, passive pattern recognition systems learn the distinctivecharacteristics of each work. During a training procedure, works thatare to be identified are analyzed and features of the audio and video(or both) are processed into templates to be recognized later. Unknowninput signals are then analyzed and compared with the features of eachknown pattern. When the properties of the unknown audio or video signalmatch one of the template sets stored in a database, the unknown inputis declared to match the work that was used to produce the correspondingtemplates. This is analogous to fingerprint or DNA matching. By properlyselecting the features of the audio or video that are used to constructthe stored templates this process can be extremely reliable, even incases where the signal has been significantly degraded and distorted.The system can of course learn to recognize any work, old or new.

In most implementations of passive signal recognition technology, thetemplates stored in the database are derived from a single time intervalof a recording that may range from several seconds to a minute induration. The system then monitors each input cannel continuously,searching for a match with one of the templates in the database. In thisconfiguration the system has only learned a small piece of each workthat it must recognize. As the system searches for audio or videopattern matches on its input channels it must repeatedly acquire signalsegments and compare them with database entries. The system mustcontinuously monitor each of its input channels. Otherwise, a timesegment that matches one of the database templates could occur when thesystem is not monitoring a particular channel.

A system based on the present invention is designed differently. Insteadof learning a single time segment from each audio or video work, all ofthe time segments comprising each work are learned. While this increasesthe size of the pattern database, the size is not unreasonable. Signalrecognition is accomplished from a single input signal segment. Once aninput segment has been captured, it is compared with all storedtemplates from all monitored works. The signal input stream appearing ata particular input port can then be switched to a different channel.This multiplexing or channel hopping can be done without fear of missinga detection so long as the system revisits each channel within theduration of a particular work. If a segment is missed because the systemis observing a different channel, the audio or video work will beidentified by matching a later time segment when the system switchesback to the proper channel. This procedure is analogous to what a humanobserver might do if he were to try to keep track of the program contentof many television channels using a single receiver. Assuming that theobserver knew all of the programs that could possibly be transmitted, hecould identify the program on one channel or information stream and thenswitch to a different channel and identify that program as well. Thisprocedure can be repeated for many channels or Internet virtual channelswithout risk that a program will be missed.

The present signal recognition method is also able to identify briefclips or excerpts from registered programming. Further, since the systemhas learned the entire program it is able to determine the point in timein the program from which the excerpt was extracted. This informationcan be used to determine whether a program has been transmitted in itsentirety or if it has been edited to remove certain portions. The systemarchitecture is also capable of detecting programs that have beenconstructed by splicing together portions of several other copyrightprotected works. Again, since the system will know all of the availableprogramming it is able to indicate which time intervals of each originalwork have been extracted to produce a new program. Similarly, ifinsertions have been made into a program for advertisements or otherbreaks in continuity, this can be detected by measuring the timeintervals between program segments.

The system architecture is a distributed network of specially equippedcomputers. This network can grow in a uniform way to expand the numberof monitored channels or the number of audio or video signals to beidentified. Signal sources include Internet distribution of audio andvideo recordings, satellite downlinks that are used for broadcast feeds,or direct terrestrial and satellite distribution to consumers.Regardless of the signal source, the pattern recognition processesinvolved are the same. Separate interfaces can be provided between thesesignal sources and the signal recognition system. The design of thesystem supports growth and reconfiguration to support changing needs.

One of the initial applications of the subject program identificationsystem is to monitor computer network distribution of copyrightprotected audio and video material. These sources would include musicand video on demand services and real-time Internet broadcast of audioand video. The result of this monitoring is a set of files that indicatewhich sites are transmitting specific titles. This information can thenbe cross-indexed to determine which sites are licensed to transmit thesespecific works. In cases where there is an apparent copyrightinfringement, the appropriate rights organization can be notified sothat royalties can be collected in accordance with copyright laws andinternational agreements.

The present invention requires an initialization or registration processto produce templates of works that are later to be identified. In thisprocess, audio and video signals are digitized and processed to extractsequences of important features. For audio signals these features may bemeasurements of energy present in different portions of the audiospectrum. Video signals may be characterized by measurements of theintensity, color, texture, etc. taken from different regions of theviewing area. In both the audio and video cases, sequences of thesemeasurements constitute time series data streams that indicate thedynamic structure of the signal. For the purposes of this invention theaudio and video features are treated identically, allowing the mostdescriptive features to be used to construct the templates. The multiplefeature streams are then broken into overlapping time intervals orsegments of several seconds each that cover the entire work. The audioand/or video features from each segment are then analyzed to determinewhich features carry the most descriptive information about the segment.Features are then rank ordered according to their information content,and the best features are selected to construct a template of aparticular segment. Note that each segment may use a different subset ofavailable features, and they may be ordered differently within eachsegment. The features are then normalized and fast Fourier transformedto produce complex spectra that facilitate fast feature correlation. Inaddition, each feature is correlated with a large number of similarfeatures stored in the pattern library. This allows us to estimate thestandard deviation of correlation values when the segment is not presentin the input stream. From this we can predict the probability that aparticular peak correlation value occurred randomly. The rank orderedfeatures, normalization factors, and feature standard deviations arestored as structured records within a database entry representing theentire work.

The signal recognition process operates on unknown audio and videosignals by extracting features in the same manner as was done in theinitialization process. However, instead of capturing the entire work,it is only necessary to acquire a single snapshot or time interval equalin duration to that of a template segment. All available features arecomputed from the unknown input segment. For each time segment of eachpattern in the database the most descriptive feature is correlated withthe corresponding feature measurement from the unknown input signal.Based on the peak value of the correlation function and the standarddeviation of background correlations computed during initialization, anestimate is made of the probability that the correlation occurredrandomly. If the probability is low enough, the pattern is placed on acandidate list. Patterns on the candidate list are then furtherevaluated by correlating the next most valuable feature of each patternsegment on the candidate list with the corresponding features of theunknown input. The probability of random (false) correlation is thenestimated for this feature as well. Assuming statistical independence ofthe two feature correlations, the probability that the two eventshappened randomly is the product of the individual probabilities. Thisprocess is repeated using additional features until the probability thata detection event occurred at random is low enough that there isconfidence that the detection is legitimate. Patterns on the candidatelist that exceed the probability of false detection threshold aredeleted. This iterative process of evaluating additional featuresresults in a drastic reduction in the computational load. For example,assume that for each feature correlation only five percent of thecandidate patterns produce false alarm probabilities below the thresholdfor further consideration. Then 95% of the candidates will bedisregarded on each feature correlation pass. If we use four features,the total number of correlations N_(c) that must be computed isN _(c)=(1+0.05+(0.05)²+(0.05)³)*N _(p)where N_(p) is the total number of patterns in the database. In thiscase N_(c)=1.052625*N_(p). The use of four features requires onlyslightly more computation than a single feature. By comparison, if thisiterative rejection of candidates was not used N_(c)=4*N_(p)correlations would have been required. The savings in computation issubstantial, and increases as more features are used. This allows thesystem to search for more patterns or to monitor more channels using thesame computational resources.

The sampling strategy employed involves selecting the time betweensamples in accordance with the duration of each individual work. Thesystem must search for brief advertisements or jingles almostcontinuously. However, the system can search for longer duration workssuch as movies or television programs much less frequently, perhapsevery few minutes. The required sampling interval for each pattern isstored in the pattern database. An intelligent scheduling algorithm thendetermines which patterns to correlate on each update cycle. Thescheduler also tracks sequential time segments of works that have beendetected. Once a work has been identified the pattern recognitionprocess can be focused on the expectation that the next time segment ofthat work will appear on a particular channel. As long as thisexpectation is met there is no need to commit computing resources to theconsideration of any other candidate patterns. In this situation thesystem operates in a tracking mode instead of a search mode. The systemcan then apply the correlator computing resources to other inputchannels. The scheduler thus has the capability of greatly increasingthe capacity of the system.

3. Pattern Recognition Algorithm Description

The pattern recognition algorithm is based on computing crosscorrelation functions between feature time series data extracted fromthe input signal and reference patterns or templates derived from thesignal to be identified. The performance of the correlation function isdetermined by the amount of information contained in the pattern. Ifthere is too little information in the pattern, it will have a highfalse alarm rate due to random correlations exceeding the detectionthreshold. If there is too much information in the pattern, smallvariations or distortions of the input signal will degrade the value ofthe correlation peak causing detections to be missed. For our purposesthe information content of a pattern is equal to its time-bandwidthproduct. We have found that a time-bandwidth product of 80-100 provideslow false alarm rates while still being tolerant of distortion typicalin a broadcast environment. A pattern duration of 10 seconds wouldtherefore need a bandwidth of 8-10 Hz to produce the desiredperformance. This bandwidth can be from a single information stream orfrom several separate streams with less bandwidth, provided that theindividual streams are statistically independent. Similarly, one can useseveral time segments of low bandwidth to produce the needed timebandwidth product.

The correlation function or matched filter response can be implementedin the time domain by integrating the products of time series samples ofthe template and a corresponding number of samples of the unknown inputseries and then properly normalizing the result. However, the processmust be repeated for each time delay value to be evaluated. Thecomputational load is not acceptable. A better technique known as fastconvolution is used that is based on the fast Fourier transformalgorithm. Instead of directly computing each correlation value, anentire block of correlation values is computed as the inverse Fouriertransform of the cross-power spectrum of the template time series and ablock of input data samples. The result must be normalized by theproduct of the standard deviations of the input and the template.Furthermore, if correlations are to be computed continuously thetemplate or reference pattern must be padded with zeros to double itslength and the input data must be blocked into double length buffers.This process is repeated using overlapped segments of the input data andevaluating the values of the first half of the resulting correlationfunction buffers. This method requires that the input stream bemonitored continuously. Any occurrence of the reference pattern in theinput stream will be detected in real time.

The method used in the present invention is a variation of the fastcorrelation approach where the roles of template and input data arereversed. In this approach a sequence of overlapped data buffers areacquired from the entire audio or video time series to be recognizedduring the initialization process. A set of templates is then created asthe fast Fourier transform of the normalized data buffers. As is wellknown in signal recognition technology, a post correlation normalizationis required to adjust for the signal power present in the portion of thetemplate where the input block occurs. To accomplish this a set of RMSamplitude values is computed for each of the possible time delays. Thesevalues are computed and stored in the pattern data structure duringinitialization.

In the recognition process a block of feature data is acquired from theinput stream and normalized to a fixed total power. It is then zerofilled to double its length and Fourier transformed to produce a complexspectrum. The input spectrum is then vector multiplied by each of thetemplate spectra. The resulting cross power spectra are then inverseFourier transformed to produce a set of correlation functions. These rawcorrelation functions are then normalized by dividing each value in thecorrelation by the corresponding RMS value stored in the pattern datastructure. The correlation values range from 1.0 for a perfect match to0.0 for no match to −1.0 for an exact opposite. Further, the mean valueof these correlations will always be 0.0. By computing correlationfunctions for multiple features and combining them according to theirstatistical properties we have devised an efficient and accurate methodof recognizing multivariate time series waveforms. Note that in thisalgorithm it is only necessary to acquire a single block of input data.Continuous monitoring is not required, allowing the receiver to beswitched to a different channel. Further, since we know which templateof the set produced the detection, we know how much time is remaining inthe detected audio or video. This information can be used in schedulingwhen to revisit a particular channel.

The method of the present invention uses multiple feature streamsextracted from the audio, video or both. This allows the templategeneration and the recognition process to be tailored to the mostdistinctive aspects of each recording. In addition, the patternrecognition process is staged to conserve processing capacity. In thisapproach, an initial classification is performed using only one or twofeatures. For each feature correlation that is evaluated within aparticular time segment the system estimates the probability that suchan event could occur randomly. Candidate patterns with a low probabilityof random occurrence are examined further computing the correlation withan additional feature. Correlation peaks are matched within a timewindow and the probability that the new feature correlation occurredrandomly is estimated. The system then computes the probability ofsimultaneous random correlation as the product of the individualprobabilities (assuming statistical independence). If this jointprobability is below a predetermined detection threshold, it isdetermined that the event represents a valid recognition and a detectionis logged. If the joint probability is above a separate predeterminedrejection threshold, the event is deemed to be a false alarm and thepattern is no longer considered a candidate for recognition. Otherwisean additional feature correlation is computed and the joint probabilityis updated to include the new feature information. This process isrepeated until a decision has been made or all features have beenevaluated. The basis for relating correlation values to probabilities isthe standard deviation of feature correlations between pattern templatesand a large database of similar features extracted from different works.This is performed during initialization of each work. Since thesecorrelations have approximately a normal distribution, the cumulativedistribution function can be used to estimate the probability that aparticular correlation value occurred randomly.

The implementation of the pattern recognition algorithm is intended foruse in a channel hopping environment. A set of computer controlledreceivers can be used to monitor many channels by using appropriatescheduling. The recognition process does not need to run in real time.Feature blocks can be tagged with their channel number and time andstored for later processing. However, real time detection data is usefulfor scheduling channel selections.

4. Pattern Database Organization

The pattern recognition system is driven to a large degree by thestructure of the pattern database. In order to support a variety ofoperational modes and signal types, a pattern data structure has beendevised that is hierarchical and self descriptive. As mentionedpreviously, we believe that the best pattern recognition approach is torecognize a single sample of the incoming signal by comparing it withall samples of a particular audio or video recording. When any segmentof a recording is recognized, a detection is declared and logged, and aninput port can be released to search other channels. Similarly, if noneof the pattern segments comprising a recording are identified, one canbe assured that the recording is not present and the system can switchto a different channel to acquire a sample. Continuous monitoring ofeach channel is not required It is only necessary to revisit eachchannel at an interval shorter than the recording. This is particularlyimportant in cases where, for example, a two hour movie can beidentified from a sample that is only a few seconds in duration.

Since the system must be capable of identifying audio, video, or acombination of the two a generalized representation of feature streamshas been devised that allows the most effective features to be used foreach segment. Other segments of the same recording may use completelydifferent feature sets.

One aspect that is common to all features is that they are representedas a time series of measurements of certain characteristics of the audioand video. Examples of these measurements are energy in a particularaudio band, intensity, color, and to texture (spatial frequency) of aregion of the video.

A reference pattern is structured as a three layer hierarchy. At the toplevel the pattern identification code and pattern type are indicated inthe first two words. The third word indicates the number of timesegments in the pattern. Next is a list of pointers to segmentdescriptor blocks followed by a checksum to assure block integrity.

Each segment descriptor block carries forward the pattern identificationcode and the pattern type as the first two words in the block header.Next is the segment number indicating which time interval isrepresented. The fourth word indicates the number of features in thecurrent segment block. Next is a list of pointers to feature data blocksfollowed by a checksum.

The third level in the hierarchy is the feature data block level. Inaddition to header information these blocks actually contain patternfeature data. The first three words carry the pattern identificationcode, pattern type and the segment number as was the case in the segmentdescriptor block. The fourth word in the feature data block indicatesthe feature type. The feature type word is used to select which featurestream from the input is to be compared with this block. Next is a scalefactor that is used to adjust the relative gain among features tomaintain precision. This is necessary since the feature time series dataare normalized to preserve dynamic range. The standard deviation ofbackground (false alarm) correlations is stored along with detection andrejection probability thresholds. Next in the feature data block is afrequency domain matched filter derived from the normalized featuredata. The feature normalization array is stored next in compressed form.At the end of the block is a checksum, again to assure data structureintegrity.

In addition to the signal feature data structures that are stored in thereference pattern database are a set of structures that provideinformation about the work itself such as the name, type, author, andpublisher of each work and various industry standard identificationcodes such as ISWC, ISRC, and ISCI. Also included in this structure arethe media source type, work duration, and the date and time of patterninitialization. These structures are indexed by the same Pattern ID codeused to reference the signal feature data structures. The workdescription data are used in report generation to provide informationthat is useful to users.

5. The Structure

The structure of an automatic signal recognition system according to thepresent invention is shown in FIG. 1. This audio and video recognitionstation preferably comprises one or more Audio/Video InterfaceSubsystems 1 which accept input signals that are to be identified fromvarious sources. Each subsystem processes audio and video signals andextracts important characteristics (known as features) from thesesignals. Many signal sources can be processed simultaneously in each ofthese subsystems, and many interface subsystems can be included in thesystem structure to accommodate any number of input channels. Forexample, in a large city, enough interface subsystems may be provided tomonitor all broadcast and cable TV stations, and all AM and FM radiostations within that city. Internet host sites can be monitored fromanywhere in the world.

The Audio/Video Interface Subsystem 1 operates under the command of theSearch Robot and Scheduler Subsystem 5. The Scheduler determines whichof the input sources (e.g., TV station) needs to be sampled at which (orany) point in time to acquire feature packets for identification. Thisallows sharing of input channels among a larger number of signal sources(channel hopping) according to whether the program material from aparticular source has already been identified. The feature packetsproduced by the Audio/Video Interface Subsystems (to be described inmore detail below) contain low bandwidth time series waveforms of allavailable measurements of the source (audio, video, or both). Note thatin addition to the direct media source inputs, signals are alsocollected from sources such as the Internet 7 to support monitoring ofvirtual broadcasts and digital downloads.

The feature packets are then transmitted over the local network to thePattern Recognition Subsystems 2. These subsystems each compare theunknown feature packets with reference patterns from a portion of theMaster Pattern Database 4 in a manner to be described below. Theprocessing capacity of each Pattern Recognition Subsystem is large butlimited. To achieve real-time recognition of a virtually unlimitednumber of works, the Pattern Recognition Subsystems are replicated asneeded to achieve the required capacity. The detection results from eachPattern Recognition Subsystem 2 are transmitted over the local areanetwork to a Management Database System 6 that records which works aretransmitted on each source at various points in time. This informationis used to produce reports and is also used by the Search Robot andScheduler 5 to plan which sources should be sampled next by theAudio/Video Interface Subsystems 1.

The Pattern Initialization Subsystems 3 accept audio and video worksthat are to be stored in the Master Pattern Database 4. These subsystemsperform feature extraction (to be described below) in the same manner asin the real-time input processing. However, instead of constructingbrief packets of features for identification (as is done with the realtime input signals), the Initialization Subsystems 3 extract continuousfeature waveforms from the entire work. The feature waveforms are thenbroken into overlapping time-series segments and processed to determinewhich features should be stored for signal recognition and in whatorder. The resulting rank-ordered reference pattern data structures arestored in the Master Pattern Database 4. These patterns are subsequentlytransferred to the Pattern Recognition Subsystems 2 for comparison withthe unknown input feature packets.

6. Feature Extraction

The Audio/Video Interface Subsystem 1 comprises a host microcomputer anda plurality of specialized signal processor circuit boards that performthe actual feature extraction. The audio interface and signal processoraccording to the preferred embodiment is illustrated in FIG. 2. In thisexample, up to 64 audio sources can be monitored, but only eight can besimultaneously processed. Audio Input Source Select Multiplexers 8select among several audio sources for each channel. These sourceselectors are switched at a low speed as directed by the Scheduler 5.The output of each Source Select Multiplexer 8 is connected to an analogAntialias Lowpass Filter 9 to restrict the maximum audio frequency (to3.2 kHz in this example). The outputs of these filters are connected toa Channel Multiplexer 10 that rapidly scans the filter outputs. In thisexample with eight channels sampled at 8 kHz each, the ChannelMultiplexer 10 switches at a 64 kHz rate. The Channel Multiplexer outputis connected to an Analog to Digital Converter 11 that operates at theaggregate sample rate producing a multiplexed time series of theselected sources. The output of the Analog to Digital Converter 11 istransmitted to a programmable Digital Signal Processor 12 that performsthe digital processing of the audio time series waveforms to extractfeatures and construct the feature packets that are to be recognized.Digital Signal Processor 12 is a special purpose microprocessor that isoptimized for signal processing applications. It is connected to aProgram Memory 14 where programs and constants are stored and a DataMemory 13 for storage of variables and data arrays. The Digital SignalProcessor 12 also connects to the Host Computer Bus 16 using aninterface such as the PCI Bus Interface 15 for exchange of data betweenthe Digital Signal Processor and the host computer.

The audio signal processing necessary to perform the feature extractionis performed in software or firmware installed on Digital SignalProcessor 12 as depicted in FIG. 3. Digitized audio samples from one ofthe signal sources are grouped into a Sample Set 17 and merged with oneor more Previous Sample Sets 18 to form a window into the audio timeseries for periodic spectral analysis. The size of this windowdetermines the spectral resolution while the size of the new Sample Set17 determines the interval between updates. Once a block of data hasbeen prepared for analysis, it is multiplied by a function such as aHanning Window 19 to reduce the spectral leakage due to so calledend-effects caused by finite block size. The resultant time series isthen processed by a fast Fourier transform (FFT) 20 to produce thecomplex spectrum. The Power Spectrum 21 is then calculated from thecomplex spectrum by summing the squares of the real and imaginarycomponents of each frequency bin. An example of the resulting audioPower Spectrum 21 is shown in FIG. 4. This figure also indicates thepartitioning of the spectrum into several frequency bands. The totalpower in each of the frequency bands is found by integrating the powercontained in all of the frequency bins in the respective bands as shownin 22. Each time the above processes are performed, a new set of featuremeasurements generated. In most cases the update rate will still be muchhigher than desired from the point of view of feature bandwidth and theresulting data rate. For his reason, the sample rate is reduced byprocessing each frequency band feature sequence by a PolyphaseDecimating Lowpass Filter 23. In the preferred embodiment of theinvention this results in an audio feature sample rate of approximately10 Hz.

In the preferred embodiment of the invention, video signals go through adifferent set of steps to achieve feature extraction, but the resultingfeature time series waveforms are virtually identical. FIG. 5 is anillustration of the video interface and signal processing componentsthat perform these functions. Analog video sources can be chosen one ata time by the Video Input Source Select Multiplexer 24 as directed bythe Search Robot and Scheduler Subsystem 5. The selected video signal isdirected to a Video Antialias Lowpass Filter 25 to avoid distortion ofthe video signal. Since the system must accept a number of differentvideo formats with varying bandwidth, the cutoff frequency of thisfilter is programmable. The output of this filter is fed to a high speedAnalog to Digital Converter 26 with a programmable sample frequency tosupport different video formats. The video time series from the Analogto Digital Converter 26 is fed to both a Horizontal/Vertical FrameSynchronizer 27 and a Video Frame Generator 28. The Frame Synchronizer27 identifies horizontal synchronization pulses and vertical retraceintervals in the video signal and uses these to reset the horizontal andvertical address counters that define a raster scanned image. Thesecounters are contained in the Video Frame Generator 28 along withspatial averaging circuits that sum several adjacent rows and columns ofvideo pixels from the Analog to Digital Converter 26. This produces araster image with relatively low resolution. Digital video signals canalso be acquired from a Digital Video Interface 29 that receives itsinputs from either an external digital video source or from the PCI BusInterface that connects to the host microprocessor Computer Bus 35.Regardless of the signal source, Video Frame Generator 28 produces asequence of video frames in a standardized format of approximately 160by 120 pixels. These video frames are transferred sequentially to a setof dual ported Video Frame Buffer Memories 30. These memories alsoconnect to a Digital Signal Processor 32 where further spatial andtemporal processing is performed. Also connected to Digital SignalProcessor 32 are a DSP Program Memory 34 and a DSP Data Memory 31. Thesignal processor also connects to the Host Computer Bus 35 via PCI BusInterface 33.

The operations performed by Digital Signal Processor 32 to extract videofeatures are illustrated in FIG. 6. To begin the process we get the NextFrame 36 from a Video Frame Buffer Memory 30 and extract Intensity,Color, or Texture 37 from each pixel in the video frame. Atwo-dimensional Spatial Weighting Function 38 then multiplies eachregion of the video frame. An example of this weighting function isshown in FIG. 8. The approximate positions of the regions within thevideo frame are shown in FIG. 7. Note that the positions of theseregions have been selected to allow feature extraction from either 4:3or 16:9 aspect ratio video formats. After weighting the pixels in eachregion, we Integrate Over Each Region 39 by summing all of the weightedpixels in each region. Next the DSP 32 Samples Each Region 40 producinga set of feature measurements, one per region. In the example shown inFIG. 7, there are 13 feature measurements in each frame. The next stepis to Construct a Time Series for Each Region 41. A set of PolyphaseDecimating Filters is then applied to each feature time series to reducethe sample rate of each video feature to the same rate as the audio. Inthe preferred embodiment of the invention this is approximately 10 Hz.The audio and video processing boards and feature extraction processesare identical in both the Audio/Video Interface Subsystems 1 and thePattern Initialization Subsystems 3.

FIG. 9 is an example of a set of feature waveforms extracted from anaudio signal. If this had been a video only signal, there would be 13separate features. In the typical television signal case, a total of 19feature waveforms are present in the preferred embodiment of theinvention. In the case of the real-time signal recognition process, aset of 64 consecutive samples is collected from each feature waveform toconstruct recognition feature packets. In constructing referencepatterns, each feature waveform is broken into segments that are 128samples long and are overlapped by 64 samples. This ensures that anunknown input sample feature packet will be completely contained in atleast one of the feature reference segments. The overlappingsegmentation of a single feature is illustrated in FIG. 10. Thissegmentation is applied to all available features.

7. Reference Pattern Generation

The procedure for generating reference patterns is illustrated in FIG.11. For each feature of each segment, the feature waveform is firstblock-scaled to a fixed total power. This assures that the precision anddynamic range of the signal processing is preserved. The scale factorused in this scaling is saved. Next the fast Fourier transform (FFT) ofthe feature waveform is computed, yielding the complex spectrum that isused in the fast correlation algorithm. A sliding standard deviation(RMS) of the feature waveform is also computed for use in properlynormalizing the correlation functions. In the preferred embodiment ofthe invention the standard deviation is calculated for each of 64positions within a 128-sample segment using a window that is 64 sampleslong. Once all features of all segments have been processed as describedabove, the information content of each feature from each segment ismeasured.

The measure of information content used in the preferred embodiment inthe degree of spectral dispersion of energy in the power spectrum ofeach feature. This can be statistically estimated from the second momentof the power spectrum. Features with widely dispersed energy have morecomplex structure and are therefore more distinctive in their ability todiscriminate among different patterns. The features within each segmentare then rank-ordered by their information content so that the mostuseful features will be used first in the pattern recognition process.Features with too little information to be useful are deleted from thereference pattern data structure. Next, the detection parameters arecomputed. Each feature is correlated with a large number of patternsamples that do not match the pattern under consideration. Thestatistical distribution that results characterizes the false alarmbehavior of the feature. Acceptable detection and rejectionprobabilities are then computed from the joint probability of falsealarm. These are stored as detection and rejection thresholds to be usedin the pattern recognition process.

The reference pattern database structure of the preferred embodiment isillustrated in FIG. 12. This structure contains two types ofinformation, both of which are indexed by a unique PatternIdentification Code 43. The first is a descriptive data record 45 thatcontains administrative information such as the name, type, author, andpublisher of the work. Also included are various industry standardidentification codes and data that describe the source media andinitialization time and date. The pattern identification code is alsoincluded in this record to allow cross-checking the database.

The second part of the database is a hierarchical set of data structuresthat contain the reference pattern data itself plus the informationneeded to process the data. At the top of this hierarchy is the PatternDescriptor Block 44. This block contains the pattern identification codeto allow integrity checking of the database and the pattern type (audio,video, mixed, etc.). Next is a number that indicates the number ofsegments in the pattern and a set of pointers to Segment DescriptorBlocks 46. A checksum may also be included to verify the integrity ofthe block. The Segment Descriptor Blocks contain the patternidentification code, pattern type and segment number to verify theintegrity of the data structures. Next are the number of features, alist of pointers to feature blocks, and an optional checksum. EachFeature Block 47 contains the pattern identification code, pattern type(audio, video, mixed, etc.), segment number, and feature type (audio,video, etc.). Next is the scale factor that was used to block scale thefeature waveform during initialization followed by the standarddeviation of background (false) correlations that was computed from thefalse alarm correlation distribution. The detection and rejectionprobability thresholds are included next. These are used to determinewhether a detection can be confirmed, false alarm can be confirmed, orif another feature must be evaluated in order to decide. The complexspectrum of the feature data is included next, followed by the slidingstandard deviation (RMS) of the feature waveform that is used tonormalize the raw correlation functions. A checksum may also beincluded.

8. Pattern Recognition

During the pattern recognition process, the performance and capacity ofthe system can be enhanced by using information regarding priordetections and by knowing the duration of each work to be identified. Inthe preferred embodiment of the invention this is done by the SearchRobot and Scheduler 5. The search robot function is primarily used toidentify new sources of audio and video and to examine Internet downloadsites that distribute audio and video recordings. The operation of theInput Channel and Pattern Scheduler is shown in FIG. 13. It should benoted that FIG. 13 illustrates the operation of a single channel and theacquisition of signal samples for identification. The system performsthese functions on many channels simultaneously and allocates resourcesamong the channels as needed. The first decision pertains to whether thecontent of the present channel has been identified during a previousupdate cycle. If the content of the channel has been identified, thechannel can be removed from the schedule for the remaining duration ofthe detected recording. A different channel can then be selected. If thecontents of the channel are not known, the system must acquire a newinput sample block. The reference patterns stored on the PatternRecognition Subsystems 2 are organized in several lists depending ontheir duration. Works or recordings that are relatively short must bechecked on every update cycle to ensure that they are not missed.Typically, these works are 10 to 15 seconds in duration, and are usuallyjingles and advertisements. The next list contains patterns of mediumduration, where the definition of medium duration is subjective.However, for illustrative purposes we will specify that the shortestwork on this list is 120 seconds. If the feature sample block size has aduration of 6 seconds, the system must check each pattern every 20blocks. In this example of the preferred embodiment the system willprocess five percent of the medium duration pattern list on each updatecycle. Similarly, long duration patterns are checked even lessfrequently. These patterns might be derived from television programs ormovies. For purposes of illustration we can select patterns with aminimum duration of 20 minutes to be members of the long list. For asample block duration of 6 seconds, the system need only check the listevery 200 blocks. In this example of the preferred embodiment the systemonly needs to process one-half of one percent on each update cycle toensure that a work will not be missed. Upon completion recognition ofpatterns from all of these lists, detection results are reported and thenext input channel is selected. Note that three lists of patterns havebeen discussed for illustrative purposes. It should be clear that thenumber of lists of patterns used in the preferred embodiment can bematched to the distribution of pattern durations that are actually inuse. In addition, it is possible to further reduce the number ofpatterns that must be evaluated by restricting the individual patternsonly to channels where they may be expected to appear. For example, wewould not expect to find television programs or movies broadcast onradio stations, so the system can skip these patterns when evaluatingradio broadcasts.

FIG. 14 identifies the steps that are necessary to prepare a new inputfeature block for pattern recognition. The raw input feature setcomprises a set of time series waveforms representing audio and/or videosignals. First, the mean value of each feature is removed. Next, eachfeature in the input block is normalized by dividing each feature datavalue by the standard deviation calculated over the entire block. Eachnormalized feature time series is then padded with zeros to double itsduration. This is a desirable step in the fast correlation process toprevent circular time wrapping of data values from distortingcorrelation values. The fast Fourier transform (FFT) of each feature isthen computed, producing a complex spectrum.

The pattern recognition process employed in the preferred embodiment ofthe invention are illustrated in FIG. 15. When a new input feature blockis acquired it is compared with candidate patterns on one or more of thereference pattern lists. After initializing this list to access the nextpattern to be evaluated the first feature is selected from both theunknown input and the reference pattern. The cross-correlation functionis then computed. The correlation function has a value of one for aperfect match, zero for no correlation, and negative one for a perfectanti-correlation. The maximum value of the correlation function is thenfound. This correlation peak value is then divided by the standarddeviation of background (false) correlations that was found in theinitialization process to yield the number of standard deviations fromthe mean value of zero. Using Gaussian statistics we can estimate theprobability that this event occurred randomly (a false alarm). Theprocess is repeated for subsequent features at the same instant of time.The resulting probabilities of false alarm for the individual featuresare multiplied to produce a composite false alarm probability. Thecomposite probability of false alarm (PFA) is then compared with anupper limit. If the composite PFA exceeds this limit, the candidatedetection is deemed to be a false alarm and the pattern is rejected.Otherwise the composite PFA is compared with a lower limit. If thecomposite PFA is less than the lower limit, the probability that theevent is due to random events is deemed to be sufficiently low that theevent must be a legitimate pattern recognition. The detection event isthen logged along with the time and date of its occurrence and thechannel number or source. Additional information regarding the remainingtime in the recording is passed to the scheduler to allow it to makemore efficient scheduling plans. If the composite PFA is above the lowerlimit and is below the upper limit, the cause of the event is stilluncertain and requires the use of additional information from otherfeatures. This process of correlating, estimating individual featurePFA's, updating the composite PFA and evaluating the composite PFA isrepeated until a decision can be made. Note that a new pair of PFAlimits is used each time that a new feature is added. In addition, theupper and lower PFA limits for the last available feature are set to beequal to force a decision to be made. The above processes are repeatedfor all time segments of all patterns on the candidate pattern list.This could result in simultaneous detections of two or more patterns. Ifsuch simultaneous detections occur, this could indicate that one work orrecording is a composite of other initialized works.

FIG. 16 illustrates the steps in performing the fast correlationalgorithm using the complex spectra of the feature waveforms from theunknown input and an initialized reference pattern from the database.These spectra are first multiplied to produce the complex cross-powerspectrum. The inverse fast Fourier transform is then applied to thecross-spectrum to obtain a raw correlation function. The first half ofthis correlation function is then normalized by the sliding standarddeviation (RMS) previously computed during initialization and stored inthe feature structure of the pattern database. The second half of thecorrelation function represents circularly time-wrapped values that arediscarded. An example of a properly normalized feature correlation isshown in FIG. 17.

FIG. 18 illustrates how false detection probabilities can be estimatedfrom the feature correlation values and the standard deviation ofbackground (false) correlations calculated during initialization. It hasbeen found that the distribution of random correlations is approximatelynormal resulting in a probability density function resembling FIG. 18A.In the preferred embodiment of the invention, the correlation values aredivided by the standard deviation of background correlations. Thisyields the number of standard deviations from the expected value. Thecumulative distribution function shown in FIG. 18B indicates theprobability that a correlation value expressed in standard deviationswill encompass all legitimate detections. For example, if the standarddeviation of background correlations was found to be 0.3 duringinitialization and we compute a correlation value of 0.6 during patternrecognition, the correlation value is 2 standard deviations above theexpected (mean) value for all correlations. From FIG. 18B we surmisethat this correlation value is greater than 97.7 percent of all randomlyoccurring correlation values. The probability that a random correlationwill exceed this value is therefore only 2.3 percent. This isillustrated in FIG. 18C where we define the probability of false alarmfor an individual feature to be PFA=1−cdf((correlation peak)/sigma). Inthe preferred embodiment of the invention these probabilities are storedin a table for rapid lookup. Assuming statistical independence of thefeatures, the probability that simultaneous false detections of featureswill occur is simply the product of the individual probabilities offalse alarm.

The teachings of this patent describe a methodology and a system toautomatically recognize audio and video performances in an accurate andefficient manner. Those skilled in the art will recognize that there aremany obvious variations of the methodology and system structure that arealso within the scope of these teachings and the appended claims.

What is claimed is:
 1. Audio signal recognition server apparatus adaptedto receive, from a capture device, feature data that corresponds to acaptured audio sample that is less than an entire reference audio work,the recognition server apparatus comprising: interface structureconfigured to receive the sample feature data from the capture device; amemory storing a library comprising (i) a first plurality of referencefeature data sets which correspond to a first recorded reference audiowork, and (ii) a second plurality of reference feature data sets whichcorrespond to a second recorded reference audio work, each recordedreference audio work being longer than the captured audio sample; andserver processing structure configured to: receive a first referenceinput audio signal corresponding to the first recorded reference audiowork; separate the received first reference input audio signal into afirst plurality of frequency bands which have different frequencies, afrequency bandwidth of a lower frequency band of the first plurality offrequency bands being narrower than a frequency bandwidth of a higherfrequency band of the first plurality of frequency bands; compute thefirst plurality of reference feature data sets, which correspond tospectrally distinct portions of the first plurality of frequency bandsof the first received reference input audio signal, this computingcomprising performing envelope extraction on the first plurality offrequency bands to provide low-bandwidth amplitude measurements of eachof the first plurality of frequency bands to provide the first pluralityof reference feature data sets; store in the memory the first pluralityof reference feature data sets which correspond to the first referenceinput audio signal; receive a second reference input audio signalcorresponding to the second recorded reference audio work; separate thereceived second reference input audio signal into a second plurality offrequency bands which have different frequencies, a frequency bandwidthof a lower frequency band of the second plurality of frequency bandsbeing narrower than a frequency bandwidth of a higher frequency band ofthe second plurality of frequency bands; compute the second plurality ofreference feature data sets, which correspond to spectrally distinctportions of the second plurality of frequency bands of the secondreceived reference input audio signal, this computing comprisingperforming envelope extraction on the second plurality of frequencybands to provide low-bandwidth amplitude measurements of each of thesecond plurality of frequency bands to provide the second plurality ofreference feature data sets; store in the memory the second plurality ofreference feature data sets which correspond to the second referenceinput audio signal; compare the sample feature data received by saidinterface structure with the stored first and second pluralities ofreference feature data sets; and generate a recognition signal inresponse to the received sample feature data matching at least onereference feature data set of the stored first and second pluralities ofreference feature data sets.
 2. Apparatus according to claim 1, furthercomprising the capture device.
 3. Apparatus according to claim 2,wherein the capture device comprises a non-portable device.
 4. Apparatusaccording to claim 1, wherein said interface structure is configured toreceive the sample feature data at least in part via the Internet. 5.Apparatus according to claim 1, wherein the recognition signalidentifies a video work.
 6. Audio signal recognition server apparatusadapted to receive, from a capture device, feature data that correspondsto a captured audio sample that is less than an entire reference audiowork, the recognition server apparatus comprising: interface structureconfigured to receive the sample feature data from the capture device; amemory storing a library comprising (i) a first plurality of referencefeature data sets which correspond to a first recorded reference audiowork, and (ii) a second plurality of reference feature data sets whichcorrespond to a second recorded reference audio work, each recordedaudio work being longer than the captured audio sample, the firstplurality of reference feature data sets corresponding to a firstplurality of frequency bands which have different frequencies, afrequency bandwidth of a lower frequency band of the first plurality offrequency bands being narrower than a frequency bandwidth of a higherfrequency band of the first plurality of frequency bands, the firstplurality of reference feature data sets including features whichcorrespond to spectrally distinct portions of the first plurality offrequency bands, the second plurality of reference feature data setscorresponding to a second plurality of frequency bands which havedifferent frequencies, a frequency bandwidth of a lower frequency bandof the second plurality of frequency bands being narrower than afrequency bandwidth of a higher frequency band of the second pluralityof frequency bands, the second plurality of reference feature data setsincluding features which correspond to spectrally distinct portions ofthe second plurality of frequency bands; and server processing structureconfigured to: compare the sample feature data received by saidinterface structure with the stored first and second pluralities ofreference feature data sets; and generate a recognition signal inresponse to the received sample feature data matching at least onereference feature data set of the stored first and second pluralities ofreference feature data sets.
 7. Apparatus according to claim 6, whereinthe server processing structure is also configured to: receive a firstreference input audio signal corresponding to the first recordedreference audio work; separate the received first reference input audiosignal into the first plurality of frequency bands which have differentfrequencies; compute the first plurality of reference feature data sets,which correspond to spectrally distinct portions of the first pluralityof frequency bands of the first received reference input audio signal,this computing comprising performing envelope extraction on the firstplurality of frequency bands to provide low-bandwidth amplitudemeasurements of each of the first plurality of frequency bands toprovide the first plurality of reference feature data sets; store in thememory the first plurality of reference feature data sets whichcorrespond to the first reference input audio signal; receive a secondreference input audio signal corresponding to the second recordedreference audio work; separate the received second reference input audiosignal into the second plurality of frequency bands which have differentfrequencies; compute the second plurality of reference feature datasets, which correspond to spectrally distinct portions of the secondplurality of frequency bands of the second received reference inputaudio signal, this computing comprising performing envelope extractionon the second plurality of frequency bands to provide low-bandwidthamplitude measurements of each of the second plurality of frequencybands to provide the second plurality of reference feature data sets;and store in the memory the second plurality of reference feature datasets which correspond to the second reference input audio signal. 8.Audio signal recognition server apparatus comprising: interfacestructure configured to (i) receive a first reference input audio signalcorresponding to a first recorded reference audio work, and (ii) receivea second reference input audio signal corresponding to a second recordedreference audio work; a memory storing a library comprising (i) a firstplurality of reference feature data sets which correspond to the firstrecorded reference audio work, and (ii) a second plurality of referencefeature data sets which correspond to the second recorded referenceaudio work, each recorded reference audio work being longer than areceived sample signal; and server processing structure configured to:separate the received first reference input audio signal into a firstplurality of frequency bands which have different frequencies, afrequency bandwidth of a lower frequency band of the first plurality offrequency bands being narrower than a frequency bandwidth of a higherfrequency band of the first plurality of frequency bands; compute thefirst plurality of reference feature data sets, which correspond tospectrally distinct portions of the first plurality of frequency bandsof the first received reference input audio signal; store in the memorythe first plurality of reference feature data sets which correspond tothe first reference input audio signal; separate the received secondreference input audio signal into a second plurality of frequency bandswhich have different frequencies, a frequency bandwidth of a lowerfrequency band of the second plurality of frequency bands being narrowerthan a frequency bandwidth of a higher frequency band of the secondplurality of frequency bands; compute the second plurality of referencefeature data sets, which correspond to spectrally distinct portions ofthe second plurality of frequency bands of the second received referenceinput audio signal; and store in the memory the second plurality ofreference feature data sets which correspond to the second referenceinput audio signal.
 9. Apparatus according to claim 8, wherein theinterface structure is configured to receive, from a capture device,feature data that corresponds to a portion of a captured audio samplethat is less than an entire reference audio work, and wherein the serverprocessing structure is also configured to: compare the sample featuredata received by said interface structure with the stored first andsecond pluralities of reference feature data sets; and generate arecognition signal in response to the sample feature data matching atleast one feature data set of the stored first and second pluralities ofreference feature data sets.
 10. An audio signal recognition methodadapted to receive, from a capture device, feature data that correspondsto a captured audio sample that is less than an entire reference audiowork, the recognition server method comprising: receiving, with aninterface structure, the sample feature data from the capture device;storing, in a memory, a library comprising (i) a first plurality ofreference feature data sets which correspond to a first recordedreference audio work, and (ii) a second plurality of reference featuredata sets which correspond to a second recorded reference audio work,each recorded reference audio work being longer than the captured audiosample; and using a server processing structure to: receive a firstreference input audio signal corresponding to the first recordedreference audio work; separate the received first reference input audiosignal into a first plurality of frequency bands which have differentfrequencies, a frequency bandwidth of a lower frequency band of thefirst plurality of frequency bands being narrower than a frequencybandwidth of a higher frequency band of the first plurality of frequencybands; compute the first plurality of reference feature data sets, whichcorrespond to spectrally distinct portions of the first plurality offrequency bands of the first received reference input audio signal;store in the memory the first plurality of reference feature data setswhich correspond to the first reference input audio signal; receive asecond reference input audio signal corresponding to the second recordedreference audio work; separate the received second reference input audiosignal into a second plurality of frequency bands which have differentfrequencies, a frequency bandwidth of a lower frequency band of thesecond plurality of frequency bands being narrower than a frequencybandwidth of a higher frequency band of the second plurality offrequency bands; compute the second plurality of reference feature datasets, which correspond to spectrally distinct portions of the secondplurality of frequency bands of the second received reference inputaudio signal; store in the memory the second plurality of referencefeature data sets which correspond to the second reference input audiosignal; compare the sample feature data received by said interfacestructure with the stored first and second pluralities of referencefeature data sets; and generate a recognition signal in response to thereceived sample feature data matching at least one reference featuredata set of the stored first and second pluralities of reference featuredata sets.
 11. An audio signal recognition server method adapted toreceive, from a capture device, feature data that corresponds to acaptured audio sample that is less than an entire reference audio work,the recognition server method comprising: receiving, with an interfacestructure, the sample feature data from the capture device; storing, ina memory, a library comprising (i) a first plurality of referencefeature data sets which correspond to a first recorded reference audiowork, and (ii) a second plurality of reference feature data sets whichcorrespond to a second recorded reference audio work, each recordedreference audio work being longer than the captured audio sample, thefirst plurality of reference feature data sets corresponding to a firstplurality of frequency bands which have different frequencies, afrequency bandwidth of a lower frequency band of the first plurality offrequency bands being narrower than a frequency bandwidth of a higherfrequency band of the first plurality of frequency bands, the firstplurality of reference feature data sets including features whichcorrespond to spectrally distinct portions of the first plurality offrequency bands, the second plurality of reference feature data setscorresponding to a second plurality of frequency bands which havedifferent frequencies, the second plurality of reference feature datasets including features which correspond to spectrally distinct portionsof the second plurality of frequency bands, a frequency bandwidth of alower frequency band of the second plurality of frequency bands beingnarrower than a frequency bandwidth of a higher frequency band of thesecond plurality of frequency bands; and using a server processingstructure to: compare the sample feature data received by said interfacestructure with the stored first and second pluralities of referencefeature data sets; and generate a recognition signal in response to thereceived sample feature data matching at least one feature data set ofthe stored first and second pluralities of reference feature data sets.12. A method according to claim 11, wherein the server processingstructure is further used to: receive a first reference input audiosignal corresponding to the first recorded reference audio work;separate the received first reference input audio signal into the firstplurality of frequency bands which have different frequencies; computethe first plurality of reference feature data sets, which correspond tospectrally distinct portions of the first plurality of frequency bandsof the first received reference input audio signal, this computingcomprising performing envelope extraction on the first plurality offrequency bands to provide low-bandwidth amplitude measurements of eachof the first plurality of frequency bands to provide the first pluralityof reference feature data sets; store in the memory the first pluralityof reference feature data sets which correspond to the first referenceinput audio signal; receive a second reference input audio signalcorresponding to the second recorded reference audio work; separate thereceived second reference input audio signal into the second pluralityof frequency bands which have different frequencies; compute the secondplurality of reference feature data sets, which correspond to spectrallydistinct portions of the second plurality of frequency bands of thesecond received reference input audio signal, this computing comprisingperforming envelope extraction on the second plurality of frequencybands to provide low-bandwidth amplitude measurements of each of thesecond plurality of frequency bands to provide the second plurality ofreference feature data sets; and store in the memory the secondplurality of reference feature data sets which correspond to the secondreference input audio signal.
 13. An audio signal recognition servermethod comprising: receiving, with an interface structure, (i) a firstreference input audio signal corresponding to the a first recordedreference audio work, and (ii) a second reference input audio signalcorresponding to a second recorded reference audio work; storing, in amemory, a library comprising (i) a first plurality of reference featuredata sets which correspond to the first recorded reference audio work,and (ii) a second plurality of reference feature data sets whichcorrespond to the second recorded reference audio work, each recordedreference audio work being longer than a portion of a captured audiosample; and using a server processing structure to: separate thereceived first reference input audio signal into a first plurality offrequency bands which have different frequencies, a frequency bandwidthof a lower frequency band of the first plurality of frequency bandsbeing narrower than a frequency bandwidth of a higher frequency band ofthe first plurality of frequency bands; compute the first plurality ofreference feature data sets, which correspond to spectrally distinctportions of the first plurality of frequency bands of the first receivedreference input audio signal; store in the memory the first plurality ofreference feature data sets which correspond to the first referenceinput audio signal; separate the received second reference input audiosignal into a second plurality of frequency bands which have differentfrequencies, a frequency bandwidth of a lower frequency band of thesecond plurality of frequency bands being narrower than a frequencybandwidth of a higher frequency band of the second plurality offrequency bands; compute the second plurality of reference feature datasets, which correspond to spectrally distinct portions of the secondplurality of frequency bands of the second received reference inputaudio signal; and store in the memory the second plurality of referencefeature data sets which correspond to the second reference input audiosignal.
 14. A method according to claim 13, wherein the interfacestructure receives, from a capture device, feature data that correspondsto a captured audio sample that is less than an entire reference audiowork, and wherein the server processing structure is further used to:compare the sample feature data received by said interface structurewith the stored first and second pluralities of reference feature datasets; and generate a recognition signal in response to the receivedsample feature data matching at least one feature data set of the storedfirst and second pluralities of reference feature data sets.
 15. Atleast one computer readable non-transitory medium for one or more audiosignal recognition servers which are adapted to receive, from a capturedevice, feature data that corresponds to a captured audio sample that isless than an entire reference audio work, the at least one computerreadable medium having instructions which, when read by one or moreprocessing structures of the one or more recognition servers, cause theone or more processing structures to: store, in a memory, a librarycomprising (i) a first plurality of reference feature data sets whichcorrespond to a first recorded reference audio work, and (ii) a secondplurality of reference feature data sets which correspond to a secondrecorded reference audio work, each recorded reference audio work beinglonger than the portion of the captured audio sample; receive a firstreference input audio signal corresponding to the first recordedreference audio work; separate the received first reference input audiosignal into a first plurality of frequency bands which have differentfrequencies, a frequency bandwidth of a lower frequency band of thefirst plurality of frequency bands being narrower than a frequencybandwidth of a higher frequency band of the first plurality of frequencybands; compute the first plurality of reference feature data sets, whichcorrespond to spectrally distinct portions of the first plurality offrequency bands of the first received reference input audio signal;store in the memory the first plurality of reference feature data setswhich correspond to the first reference input audio signal; receive asecond reference input audio signal corresponding to the second recordedreference audio work; separate the received second reference input audiosignal into a second plurality of frequency bands which have differentfrequencies; a frequency bandwidth of a lower frequency band of thesecond plurality of frequency bands being narrower than a frequencybandwidth of a higher frequency band of the second plurality offrequency bands compute the second plurality of reference feature datasets, which correspond to spectrally distinct portions of the secondplurality of frequency bands of the second received reference inputaudio signal; store in the memory the second plurality of referencefeature data sets which correspond to the second reference input audiosignal; compare the sample feature data received from the capture devicewith the stored first and second pluralities of reference feature datasets; and generate a recognition signal in response to the receivedsample feature data matching at least one feature data set of the storedfirst and second pluralities of reference feature data sets.
 16. Atleast one computer readable non-transitory medium for one or more audiosignal recognition servers which are adapted to receive, from a capturedevice, feature data that corresponds to a captured audio sample that isless than an entire reference audio work, the at least one computerreadable medium having instructions which, when read by one or moreprocessing structures of the one or more recognition servers, cause theone or more processing structures to: store, in a memory, a librarycomprising (i) a first plurality of reference feature data sets whichcorrespond to a first recorded reference audio work, and (ii) a secondplurality of reference feature data sets which correspond to a secondrecorded reference audio work, each recorded reference audio work beinglonger than the captured audio sample, the first plurality of referencefeature data sets corresponding to a first plurality of frequency bandswhich have different frequencies, a frequency bandwidth of a lowerfrequency band of the first plurality of frequency bands being narrowerthan a frequency bandwidth of a higher frequency band of the firstplurality of frequency bands, the first plurality of reference featuredata sets including features which correspond to spectrally distinctportions of the first plurality of frequency bands, the second pluralityof reference feature data sets corresponding to a second plurality offrequency bands which have different frequencies, a frequency bandwidthof a lower frequency band of the second plurality of frequency bandsbeing narrower than a frequency bandwidth of a higher frequency band ofthe second plurality of frequency bands, the second plurality ofreference feature data sets including features which correspond tospectrally distinct portions of the second plurality of frequency bands;compare the sample feature data received by said interface structurewith the stored first and second pluralities of reference feature datasets; and generate a recognition signal in response to the receivedsample feature data matching at least one feature data set of the storedfirst and second pluralities of reference feature data sets.
 17. Acomputer readable non-transitory medium according to claim 16, whereinthe at least one computer readable medium instructions cause the one ormore processing structures to: receive a first reference input audiosignal corresponding to the first recorded reference audio work;separate the received first reference input audio signal into the firstplurality of frequency bands which have different frequencies; computethe first plurality of reference feature data sets, which correspond tospectrally distinct portions of the first plurality of frequency bandsof the first received reference input audio signal, this computingcomprising performing envelope extraction on the first plurality offrequency bands to provide low-bandwidth amplitude measurements of eachof the first plurality of frequency bands to provide the first pluralityof reference feature data sets; store in the memory the first pluralityof reference feature data sets which correspond to the first referenceinput audio signal; receive a second reference input audio signalcorresponding to the second recorded reference audio work; separate thereceived second reference input audio signal into the second pluralityof frequency bands which have different frequencies; compute the secondplurality of reference feature data sets, which correspond to spectrallydistinct portions of the second plurality of frequency bands of thesecond reference received input audio signal, this computing comprisingperforming envelope extraction on the second plurality of frequencybands to provide low-bandwidth amplitude measurements of each of thesecond plurality of frequency bands to provide the second plurality ofreference feature data sets; and store in the memory the secondplurality of reference feature data sets which correspond to the secondreference input audio signal.
 18. At least one computer readablenon-transitory medium for one or more audio signal recognition serverswhich are adapted to receive, from a capture device, feature data thatcorresponds to a captured audio sample that is less than an entirereference audio work, the at least one computer readable medium havinginstructions which, when read by one or more processing structures ofthe one or more recognition servers, cause the one or more processingstructures to: receive, at the one or more recognition servers: (i) afirst reference input audio signal corresponding to the a first recordedreference audio work, and (ii) a second reference input audio signalcorresponding to a second recorded reference audio work; storing, in amemory, a library comprising (i) a first plurality of reference featuredata sets which correspond to the first recorded reference audio work,and (ii) a second plurality of reference feature data sets whichcorrespond to the second recorded reference audio work, each recordedreference audio work being longer than a portion of the captured audiosample; separate the received first reference input audio signal into afirst plurality of frequency bands which have different frequencies, afrequency bandwidth of a lower frequency band of the first plurality offrequency bands being narrower than a frequency bandwidth of a higherfrequency band of the first plurality of frequency bands; compute thefirst plurality of reference feature data sets, which correspond tospectrally distinct portions of the first plurality of frequency bandsof the first received reference input audio signal; store in the memorythe first plurality of reference feature data sets which correspond tothe first reference input audio signal; separate the received secondreference input audio signal into a second plurality of frequency bandswhich have different frequencies, a frequency bandwidth of a lowerfrequency band of the second plurality of frequency bands being narrowerthan a frequency bandwidth of a higher frequency band of the secondplurality of frequency bands; compute the second plurality of referencefeature data sets, which correspond to spectrally distinct portions ofthe second plurality of frequency bands of the second received referenceinput audio signal; and store in the memory the second plurality ofreference feature data sets which correspond to the second referenceinput audio signal.
 19. A computer readable non-transitory mediumaccording to claim 18, wherein the at least one computer readable mediuminstructions cause the one or more processing structures to: receive atthe one or more recognition servers, from a capture device, feature datathat corresponds to the captured audio sample that is less than anentire reference audio work; compare the sample feature data received bysaid interface structure with the stored first and second pluralities ofreference feature data sets; and generate a recognition signal inresponse to the received sample feature data matching at least onefeature data set of the stored first and second pluralities of referencefeature data sets.